1. Field of the Invention
This invention generally relates to a method for coding a voice signal, and, in particular, to a method for compressing voice data by coding.
2. Description of the Prior Art
For example, in the case of transmitting a voice signal using a high-speed digital transmission line or subjecting a voice signal to digital processing so as to store or synthesize a voice signal for use in a voice response device, it is necessary to somehow convert a voice signal into a digital signal. A voice signal is essentially an analog signal having a frequency band ranging from 0.3 to 3.4 kHz. In order to convert such a voice signal into a digital signal, use may, for example, be made of an analog-to-digital converter having an 8-bit resolution at the sampling frequency of 8 kHz (pulse-code modulation or simply PCM coding method). And, in order to restore this digital signal into the original voice signal, use may be made of a digital-to-analog converter having an 8-bit resolution at the sampling frequency of 8 kHz to convert the digital signal into an analog signal which is then passed through a low pass filter for wave-form shaping. In this case, the larger the resolution of each of the A/D and D/A converters (and thus the bit width of the PCM coding), the higher the quality of a reproduced voice.
Such a PCM-coded voice signal has a bit rate per second (data speed) of 64 kbps, and, thus, in order to transmit a voice signal having such a high bit rate, an extremely high transmission path is required. In addition, a memory of extremely large capacity is required to store such a voice signal. Accordingly, various proposals have been made to reduce the bit rate of a voice signal. One of them is the differential PCM coding method, in which differences of PCM codes adjacent in timed sequence are formed. The differential PCM coding method utilizes the redundancy based on the correlation of voice waveforms. Variations in value between adjacent samples fall in most cases in a limited dynamic range, the number of bits per sample can be reduced. In the adaptive differential pulse-code modulation coding method according to the CCITT recommendations, which is an improvement over the differential PCM coding method, the bit rate of 32 kbps has been realized.
Other proposed methods include the adaptive prediction coding method with adaptive bit allocation (APC-AB) which utilizes the non-standing characteristic of a voice signal and the linear prediction capability and the line spectrum pair (LSP) method which is based on a voice analysis/synthesis procedure. However, these adaptive PCM, APC-AB and LSP coding methods are very complicated in their coding and decoding processes, and a device for implementing such coding and decoding processes tends to be extremely expensive.
On the other hand, there is the quasi instantaneous companding method as one of high quality PCM voice transmission methods for use in telecommunication satellites. In accordance with the quasi instantaneous companding method, PCM-coded voice data are divided into blocks, each having a predetermined number of data, in timed sequence, and a scale data representing the most significant bit which corresponds to the maximum value in absolute value of a signal in each of the blocks is identified, followed by a step of forming a code data from a predetermined number of data containing the most significant bit. The quasi instantaneous companding method is relatively simple in the process of coding, and the bit number of one sample can be reduced with ease. However, the quasi instantaneous companding method is not satisfactory in efficiency.
Under the circumstances, as a possible method for improving the efficiency of the quasi instantaneous companding (compression and expansion) method, a combination of the differential PCM coding method and the quasi instantaneous companding method is conceivable. However, in general, if the quasi instantaneous companding method were simply applied to the differential PCM coding method, lost bits during compression would cause a transmission error, so that such an error would accumulate at the integrator of the receiver, thereby leading to a reception impossible condition. Described more in detail in this respect, let us consider the case in which a voice signal shown in FIG. 11a is to be coded according to a coding method which is defined by applying quasi instantaneous companding to the differential PCM coding method. In the first place, for the differential PCM coding, this voice signal is sampled, for example, at the sampling frequency of 8 kHz and differential values between the samples are defined. Here, a differential value between the adjacent samples is represented by an 8-bit data having a sign, i.e., an 8-bit data in the representation of 2's complement. And, under the quasi instantaneous companding conditions, one block is formed by eight samples and the transmission data per sample contains three bits. In addition, the scale data contains three bits.
Here, it is assumed that there has been obtained differential values for these eight samples #1 through #8 as shown in FIG. 12a. Within this block, a maximum in absolute value among the differential values is sample #1, so that the scale position POS in this case is determined as the most significant bit in the bit pattern of sample #1, which is bit 4. Thus, the value of the scale position POS becomes (100).sub.2. Accordingly, the transmission bits of each sample (transmission data or code data) include three bits of data from bit 5, which is located one bit higher than the scale position POS and indicates a sign (sign bit), to bit 3, i.e., bits 5, 4 and 3. As a result, in this block, a transmission data (code data) formed by arranging the scale position POS at the beginning and then the transmission bits of samples #1 through #8 in sequence has a structure as shown in FIG. 12b.
When decoding such a coded data, in the first place, the coded data of one block is decomposed three bits by three bits, and the scale position POS is identified by the first three bits. And, then, when expanding the following 3-bit coded data into an 8-bit data, the MSB bit of the coded data is arranged at a bit position which is one bit higher than the scale position POS with the value of the sign bit set in each of the bits higher than the MSB bit and with "0" set in each of the bits lower than the LSB bit. As a result, there is obtained the decoded data as shown in FIG. 12c. A comparison of this decoded data with the data prior to coding indicates the fact that the information of those bits which are lower than the transmission bits (d.c. component) is lost in the decoded data (see FIG.. 11b). That is, a loss of bits of information has taken place.
When a voice signal is reproduced on the basis of such a coded data with a loss of bits of information, a negative d.c. shift takes place due to accumulation of errors corresponding to the amount of lost bits, as shown by the one-dotted line in FIG. 11c, thereby producing a waveform which is shifted in position downward to the right as compared with the original waveform shown by the dotted line in FIG. 11c. As a result, information cannot be reconstructed properly. As one method to cope with this situation, there has been proposed "Differential Companding PCM (DC-PCM) Due To Accumulation of Lost Bits", Takahashi et al., Transactions of Electronics Communication Society, '84/10, vol. J67-B, No. 10. However, this proposed method is effective in compressing differential data in the order of 15 bits into compressed data in the order of 8 bits, but cannot be applied to a low bit rate coding method for compressing differential data in the order of 8 bits into compressed data in the order of 3 bits. That is, in the case of such a low bit rate, if the amplitude of a voice waveform changes significantly between blocks, there is a case when the scale position changes significantly between blocks. For this reason, there is a case when the accumulated error signal becomes larger in value than the effective data to be transmitted. In such a case, the data to be transmitted is overshadowed by the error signal, so that a proper data transmission cannot be carried out.